Hi there i have an asterisk box running ok, and now i am trying to integrate it with my local analog pbx So far, i have connected the fxo port of my * to an analog extension port of my analog pbx. As far as i know, if a call an extension of my analog pbx on a sip phone ( i have done the right dial plan for routing these calls to de zap channel ) the analog pbx extension should ring ... am i right? asterisk says the call is done, but the analog extension keeps in silence .. :? any clue, am i doing something wrong? Best regards.
Hi Julio. It would be nice if you show the extensions.conf that handles that kind of calls. You can do something like this: [macro-analogpbx] exten => s,1,Cut(ChannelType=CHANNEL,/,1) //check if the call comes from other Zap ch exten => s,2,GotoIf($[${ChannelType} = Zap] ? 3 : 6) //If does, go 3, othewise 6 exten => s,3,Flash() exten => s,4,SendDTMF(${analogprefix}${num}) //send the DTMF for the extension dialed exten => s,5,Hangup() exten => s,6,Dial(Zap/g${analoggroup}/${analogprefix}${num}) //if the call comes from SIP or IAX then execute Dial trough some group in zapata exten => s,7,Hangup() You can see some variables i just use for administration of my PBX, but i hope you understand the concept. Good Look - moy On 5/3/05, Julio Saura <julio.saura@dbs.es> wrote:> Hi there > > i have an asterisk box running ok, and now i am trying to integrate it > with my local analog pbx > > So far, i have connected the fxo port of my * to an analog extension > port of my analog pbx. > > As far as i know, if a call an extension of my analog pbx on a sip phone > ( i have done the right dial plan for routing these calls to de zap > channel ) the analog pbx extension should ring ... > > am i right? > > asterisk says the call is done, but the analog extension keeps in > silence .. :? > > any clue, am i doing something wrong? > > Best regards. > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-- "Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org"
Hello all is right, the analog extension should ring, but maybe your dialplan is not correct or you call a bad extension in you PBX. can you post your dialplan?, to see it. regards ----- Original Message ----- From: "Julio Saura" <julio.saura@dbs.es> To: <asterisk-users@lists.digium.com> Sent: Tuesday, May 03, 2005 2:37 PM Subject: [Asterisk-Users] asterisk to analog pbx> Hi there > > i have an asterisk box running ok, and now i am trying to integrate it > with my local analog pbx > > So far, i have connected the fxo port of my * to an analog extension > port of my analog pbx. > > As far as i know, if a call an extension of my analog pbx on a sip phone > ( i have done the right dial plan for routing these calls to de zap > channel ) the analog pbx extension should ring ... > > am i right? > > asterisk says the call is done, but the analog extension keeps in > silence .. :? > > any clue, am i doing something wrong? > > Best regards. > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
Hi i posted it this morning i guess is a asterisk@home problem... installing a new OS with * from scratch it does not even call outside connecting fxo to pots :? El mi?, 04-05-2005 a las 09:55 +0200, Mehdi Chouikh escribi?:> Hello > all is right, the analog extension should ring, but maybe your dialplan is > not correct or you call a bad extension in you PBX. > can you post your dialplan?, to see it. > regards > ----- Original Message ----- > From: "Julio Saura" <julio.saura@dbs.es> > To: <asterisk-users@lists.digium.com> > Sent: Tuesday, May 03, 2005 2:37 PM > Subject: [Asterisk-Users] asterisk to analog pbx > > > > Hi there > > > > i have an asterisk box running ok, and now i am trying to integrate it > > with my local analog pbx > > > > So far, i have connected the fxo port of my * to an analog extension > > port of my analog pbx. > > > > As far as i know, if a call an extension of my analog pbx on a sip phone > > ( i have done the right dial plan for routing these calls to de zap > > channel ) the analog pbx extension should ring ... > > > > am i right? > > > > asterisk says the call is done, but the analog extension keeps in > > silence .. :? > > > > any clue, am i doing something wrong? > > > > Best regards. > > > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
Well , problem solved the problem was with asterisk@home i have installed an asterisk from scratch and everything works fine now .. weird ./ Thanks! El mi?, 04-05-2005 a las 10:23 +0200, Julio Saura escribi?:> Hi > i posted it this morning > > i guess is a asterisk@home problem... installing a new OS with * from > scratch > > it does not even call outside connecting fxo to pots :? > > > > > > El mi?, 04-05-2005 a las 09:55 +0200, Mehdi Chouikh escribi?: > > Hello > > all is right, the analog extension should ring, but maybe your dialplan is > > not correct or you call a bad extension in you PBX. > > can you post your dialplan?, to see it. > > regards > > ----- Original Message ----- > > From: "Julio Saura" <julio.saura@dbs.es> > > To: <asterisk-users@lists.digium.com> > > Sent: Tuesday, May 03, 2005 2:37 PM > > Subject: [Asterisk-Users] asterisk to analog pbx > > > > > > > Hi there > > > > > > i have an asterisk box running ok, and now i am trying to integrate it > > > with my local analog pbx > > > > > > So far, i have connected the fxo port of my * to an analog extension > > > port of my analog pbx. > > > > > > As far as i know, if a call an extension of my analog pbx on a sip phone > > > ( i have done the right dial plan for routing these calls to de zap > > > channel ) the analog pbx extension should ring ... > > > > > > am i right? > > > > > > asterisk says the call is done, but the analog extension keeps in > > > silence .. :? > > > > > > any clue, am i doing something wrong? > > > > > > Best regards. > > > > > > _______________________________________________ > > > Asterisk-Users mailing list > > > Asterisk-Users@lists.digium.com > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > To UNSUBSCRIBE or update options visit: > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users